[asterisk-users] SIP interconnection problem

[Robert Bielik]

Hi all,

I've setup two * servers which are SIP interconnected ala osaka/toronto from the * book (before anyone sugggests using
IAX instead, no, I NEED to have them SIP interconnected for verification/test purposes). Then I have a 
Zoiper connected to one of them via IAX (so that * will not reinvite (?)). As soon as I try to call (via Zoiper) an extension
on the other * I get a "Failed to authenticate on INVITE" on the * to which the Zoiper is registered:

   -- Accepting AUTHENTICATED call from 192.168.10.113:  << Zoiper IP
      > requested format = gsm,
      > requested prefs = (),
      > actual format = ulaw,
      > host prefs = (ulaw|alaw|gsm),
      > priority = mine
   -- Executing [010001 at users:1] Dial("IAX2/2200-12940", "SIP/010001 at 192.168.10.11") in new stack
 == Using SIP RTP CoS mark 5
   -- Called 010001 at 192.168.10.11 << Other *
[Oct 23 11:08:25] NOTICE[13576]: chan_sip.c:15031 handle_response_invite: Failed to authenticate on INVITE to '"2200" <sip:2200 at 192.168.10.77>;tag=as3e4fedb8'  << 192.168.10.77 == * for Zoiper
   -- SIP/192.168.10.11-0a1716f8 is circuit-busy
 == Everyone is busy/congested at this time (1:0/1/0)
   -- Auto fallthrough, channel 'IAX2/2200-12940' status is 'CONGESTION'
   -- Hungup 'IAX2/2200-12940' 

Why does * try to authenticate on sip:2200 at 192.168.10.77, it is IAX for crying out loud :) ? I've set canreinvite=no on
the IAX phone (not sure this has any meaning in IAX at all)

Not sure that this is root of the interconnection problem, since I then get SIP/192.168.10.11.. is circuit-busy... ?

TIA
/R